search for: analogphones

On one installation of asterisk, I have a display on the console when I have a incoming call on my zaptel card. every digit was displayed, this was great. Does anyone know how I can get this back? Thanks

Hi, I'm unable to pick up parked calls after they are transfered. I get the "transfer" message when I press # and then I'm told "701" The extension I'm dialing goes to the on hold music. I'm disconnected, I hang up, dial "701" and I see this message on the console "Everyone is busy/congested at this time" I just have the default

Hi, I have 2 asterisk boxes as Gateway, in this arrangement. (PANASONIC PBX) - [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE) everything works great, in both directions (receiving and making calls), but when i get a call on the (ANALOGPHONE), I haven't been able to transfer it to another extension of the PANASONIC PBX using the flash key. I've tried the using the t T options on

The setvar command below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid="John French" <103> mailbox="101" callwaiting=yes threewaycalling=yes

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

...AX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk box on the lan. We are running asterisk on FC3 . SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK] ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK] Well the problem goes something like this. 1) I can dial a number form the softphones and when the call is answered I can hear the user on the other end but the user can't hear me 2) I can dial a number from the analog phones (...

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

Hi, I am in a predicament and any help/pointers would be appreciated. We are using chanspy to listen in on conversations. We are doing this via a web interface. The web interface lists all the ongoing calls. We click on a call and then my local phone rings allowing me to spy on the session I clicked on. But "most" of the time, when I start listening in, the two parties that are in

...ECT working? I can't get it to work - please see my logfiles below. While using an * CVS version of late September and chan_capi-0.2.5 (I guess), it worked!!! (I know, never change a running system ... should've backuped ... etc.) Here's the setup: NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones in between the two PBX I attached my * server with a FritzPCI card to the S0-bus. I have some Homeautomation on the server and * is supposed to route the calls depending on certain conditions. Switching an incoming call using * "Dial"-Feature would block the two internal B-channels, that&...

...AX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk box on the lan. We are running asterisk on FC3 . SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK] ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK ] Well the problem goes something like this. 1) I can dial a number form the softphones and when the call is answered I can hear the user on the other end but the user can't hear me 2) I can dial a number from the analog phones...

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with